Patrick:

Is your NAT SIP aware? What should be happening here is the 200 from your asterisk box will be sent to openser, and relayed out to your polycom. The polycom should send the ACK to the address in the top Record-Route header. In this trace, that is a private IP. Your NAT needs to re-write that to the public IP of your openser box as it hits the internet. If the private IP makes it out onto the internet, you will find your polycom is sending the ACK to 10.3.1.33. When the asterisk box doesn't receive the ACK, the call will drop.

Thanks, Phil

On Dec 13, 2007 6:00 PM, Patrick Baker pbaker@adeptra.com wrote:

Internal UA's connecting to the sip proxy work fine, but as soon an External UA connects to the sip proxy via a NAT'd public IP address - the call lasts for approx 20 seconds then disconnects. From what I can see openser is attempting to respond to asterisk after the session is initiated via the public IP NAT'd address - and is unable since asterisk and openser are behind the firewall on the internal network ( ELBOW routing ). Any help getting this working would be most appreciated, I would also like to document the issue to help others that encounter similar problems as mine.

External phones:

UA -> [(public IP) Firewall (NAT to localip)] -> openser -> asterisk

Internal phones:

UA -> openser -> asterisk

#### Layout ( asterisk and openser are on the same machine )

NATD public ip = xxx.206.xxx.136 -> 10.3.1.33 phone ip = xxx.206.xxx.137 openser = 10.3.1.33 p5060 asteriskSIP = 10.3.1.34 p5060

#### Asterisk SIP Config

[general] canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5060 bindaddr=10.3.1.34 sipdebug=yes

[openser] type=friend context=default insecure=very externalnotify=yes allow=all host=10.3.1.33

#### SER LOG

Dec 13 22:15:03 phonesys-slave openser[15976]: New request - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.34:5060;transport=udp F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137

#### Asterisk LOG

*CLI> [Dec 13 22:15:03] <--- SIP read from 10.3.1.33:5060 ---> INVITE sip:500@10.3.1.34:5060;transport=udp SIP/2.0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone CSeq: 2 INVITE Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 Contact: sip:pbaker@xxx.206.xxx.137 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 254

v=0 o=- 978310717 978310717 IN IP4 xxx.206.xxx.137 s=Polycom IP Phone c=IN IP4 xxx.206.xxx.137 t=0 0 m=audio 35022 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000

<-------------> [Dec 13 22:15:03] --- (16 headers 11 lines) --- [Dec 13 22:15:03] == Using TOS bits 0 [Dec 13 22:15:03] == Using CoS mark 5 [Dec 13 22:15:03] Sending to 10.3.1.33 : 5060 (no NAT) [Dec 13 22:15:03] Using INVITE request as basis request - f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 [Dec 13 22:15:03] No user 'pbaker' in SIP users list [Dec 13 22:15:03] Found peer 'openser' for 'pbaker' from 10.3.1.33:5060 [Dec 13 22:15:03] Found RTP audio format 0 [Dec 13 22:15:03] Found RTP audio format 8 [Dec 13 22:15:03] Found RTP audio format 18 [Dec 13 22:15:03] Found RTP audio format 101 [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Found audio description format PCMU for ID 0 [Dec 13 22:15:03] Found audio description format PCMA for ID 8 [Dec 13 22:15:03] Found audio description format G729 for ID 18 [Dec 13 22:15:03] Found audio description format telephone-event for ID 101 [Dec 13 22:15:03] Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Dec 13 22:15:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Looking for 500 in default (domain 10.3.1.34) [Dec 13 22:15:03] list_route: hop: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 [Dec 13 22:15:03] <--- Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Length: 0

<------------> [Dec 13 22:15:03] -- Executing [500@default:1] Playback("SIP/xxx.206.xxx.136-090b5380", "demo-abouttotry") in new stack [Dec 13 22:15:03] Audio is at 10.3.1.34 port 47068 [Dec 13 22:15:03] Adding codec 0x4 (ulaw) to SDP [Dec 13 22:15:03] Adding codec 0x8 (alaw) to SDP [Dec 13 22:15:03] Adding codec 0x100 (g729) to SDP [Dec 13 22:15:03] Adding non-codec 0x1 (telephone-event) to SDP [Dec 13 22:15:03] <--- Reliably Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<------------> [Dec 13 22:15:04] -- <SIP/xxx.206.xxx.136-090b5380> Playing 'demo-abouttotry.slin' (language 'en') [Dec 13 22:15:04] Retransmitting #1 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:05] Retransmitting #2 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:07] Retransmitting #3 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:11] Retransmitting #4 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:15] Retransmitting #5 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:18] -- Executing [500@default:2] Dial("SIP/xxx.206.xxx.136-090b5380", "IAX2/guest@misery.digium.com/s@default") in new stack [Dec 13 22:15:18] -- Called guest@misery.digium.com/s@default [Dec 13 22:15:18] -- Call accepted by 216.207.245.8 (format gsm) [Dec 13 22:15:18] -- Format for call is gsm [Dec 13 22:15:18] -- IAX2/216.207.245.8:4569-1 is ringing [Dec 13 22:15:19] -- IAX2/216.207.245.8:4569-1 answered SIP/xxx.206.xxx.136-090b5380 [Dec 13 22:15:19] Retransmitting #6 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804 From: "pbaker" sip:pbaker@xxx.206.xxx.136;tag=CFE2AAA1-5B882804 To: sip:500@xxx.206.xxx.136;user=phone;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:500@10.3.1.34 Content-Type: application/sdp Content-Length: 333

v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

---

[Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2337 retrans_pkt: Maximum retries exceeded on transmission f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 for seqno 2 (Critical Response) [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up call f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137 - no reply to our critical packet. [Dec 13 22:15:23] -- Hungup 'IAX2/216.207.245.8:4569-1' [Dec 13 22:15:23] == Spawn extension (default, 500, 2) exited non-zero on 'SIP/xxx.206.xxx.136-090b5380' [Dec 13 22:15:24] Really destroying SIP dialog 'f7232b5-5d4b9a93-e6ff617e@xxx.206.xxx.137' Method: INVITE

#### PCAP Diagram

|Time | xxx.206.xxx.137 | 10.3.1.33 | |5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.058 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.058 | Status: 407 Proxy A |SIP: Status: 407 Proxy Authentication Required | |(5060) <------------------ (5060) | |5.140 | Request: ACK sip:50 |SIP: Request: ACK sip:500@xxx.206.xxx.136:5060 | |(5060) ------------------> (5060) | |5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.165 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |6.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |7.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |9.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |13.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |17.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |21.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) |

#### OpenSER Config 1.2

######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Thu Dec 13 22:43:43 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support@sipwise.com) ########################################################################

######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ########################################################################

######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ########################################################################

######################################################################## # Configuration 'sip:wizard - Thu Dec 13 22:43:43 +0100 2007' ########################################################################

listen = udp:10.3.1.33:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = yes check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes

loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6)

loadmodule "textops.so"

loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0)

loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0)

loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0)

loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t")

loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain")

loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)")

loadmodule "sl.so" modparam("sl", "enable_stats", 1)

loadmodule "uri.so"

loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)")

loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256)

loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1)

loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)")

loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid")

loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0)

loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid")

######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) {

            xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            sl_send_reply("513", "Message Too Big");
            exit;
    }
    if (!mf_process_maxfwd_header("10"))
    {

            xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            sl_send_reply("483", "Too Many Hops");
            exit;
    }
    if(!is_method("REGISTER"))
    {
            if(nat_uac_test("19"))
            {
                    record_route(";nat=yes");
            }
            else
            {
                    record_route();
            }
    }
    if(is_method("CANCEL") || is_method("BYE"))
    {
            unforce_rtp_proxy();
    }
    if(loose_route())
    {
            if(!has_totag())
            {

                    xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
                    sl_send_reply("403", "Initial Loose-Routing Rejected");
                    exit;
            }
            if(nat_uac_test("19") || search("^Route:.*;nat=yes"))
            {
                    fix_nated_contact();
                    setbflag(6);
            }

            route(3);
    }
    if(is_method("REGISTER"))
    {
            route(2);
    }
    if(is_method("INVITE"))
    {
            route(4);
    }
    if(is_method("CANCEL") || is_method("ACK"))
    {
            route(8);
    }

    route(9);

}

######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); }

}

######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) {

            xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            www_challenge("", "0");
            exit;
    }
    if(!check_to())
    {

            xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            sl_send_reply("403", "Spoofed To-URI Detected");
            exit;
    }
    consume_credentials();
    if(!search("^Contact:[ ]*\*") && nat_uac_test("19"))
    {
            fix_nated_register();
            setbflag(6);
    }
    if(!save("location"))
    {

            xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            sl_reply_error();
            exit;
    }

    xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
    exit;

}

######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); }

            t_on_reply("2");
    }
    else
    {

            t_on_reply("1");
    }
    if(!isflagset(21))
    {

            t_on_failure("2");
    }
    if(isflagset(29))
    {
            append_branch();
    }
    if(is_present_hf("Proxy-Authorization"))
    {
            consume_credentials();
    }

    xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
    # no 100 (we already sent it) and no DNS blacklisting
    if(!t_relay("0x05"))
    {
            sl_reply_error();
            if(is_method("INVITE") && isbflagset(6))
            {
                    unforce_rtp_proxy();
            }
    }
    exit;

}

######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) {

            xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            setflag(23);
    }
    else
    {
            if(!proxy_authorize("", "subscriber"))
            {

                    xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
                    proxy_challenge("", "0");
                    exit;
            }
            if(!check_from())
            {

                    xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
                    sl_send_reply("403", "Spoofed From-URI Detected");
                    exit;
            }
    }
    if(nat_uac_test("19"))
    {
            fix_nated_contact();
            setbflag(6);
    }

    route(5);

}

######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20);

            route(7);
    }
    if(does_uri_exist())
    {

            xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(6);
    }
    else
    {

            xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(7);
    }
    exit;

}

######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) {

            xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            sl_send_reply("404", "User Offline");
    }
    else
    {

            xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(3);
    }
    exit;

}

######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) {

            xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(3);
            exit;
    }
    if(!isflagset(23))
    {
            # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded
            if(uri =~ "^sip:[0-9]+@")
            {
                    # only route numeric users to PSTN
                    if(!load_gws())
                    {

                            xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
                            sl_send_reply("503", "PSTN Termination Currently Unavailable");
                            exit;
                    }
                    if(!next_gw())
                    {

                            xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
                            sl_send_reply("503", "PSTN Termination Currently Unavailable");
                            exit;
                    }
                    setflag(21);

                    t_on_failure("1");
                    route(3);
            }
    }

    xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
    sl_send_reply("404", "User Not Found");
    exit;

}

######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) {

            xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            if(!t_relay())
            {
                    sl_reply_error();
            }
    }
    else
    {

            xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
    }
    exit;

}

######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit;

}

######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) {

            xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(1);
            exit;
    }

}

######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit;

}

######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit;

}

######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) {

            xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
            route(1);
            exit;
    }

    t_on_failure("1");
    route(3);

}

######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); }

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